11: Multimedia Networking 11-1 Chapter 11 Multimedia Networking A note on the use of these ppt slides: We’re making these slides freely available to all.

11: Multimedia Networking 11-1

Chapter 11Multimedia Networking

A note on the use of these ppt slides:We’re making these slides freely available to all (faculty, students, readers). They’re in PowerPoint form so you can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following: If you use these slides (e.g., in a class) in substantially unaltered form, that you mention their source (after all, we’d like people to use our book!) If you post any slides in substantially unaltered form on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material.

Thanks and enjoy! JFK / KWR

All material copyright 1996-2009J.F Kurose and K.W. Ross, All Rights Reserved

Computer Networking: A Top Down Approach

5th edition. Jim Kurose, Keith Ross

Addison-Wesley, April 2009.


Multimedia and Quality of Service: What is it?

multimedia applications: network audio and video(“continuous media”)

network provides application with level of performance needed for application to function.

QoS


Chapter 7: goals

Principles classify multimedia applications identify network services applications need making the best of best effort service

Protocols and Architectures specific protocols for best-effort mechanisms for providing QoS architectures for QoS


Chapter 7 outline

7.1 multimedia networking applications

7.2 streaming stored audio and video

7.3 making the best out of best effort service

7.4 providing multiple classes of service

7.5 providing QoS guarantees


MM Networking Applications

Fundamental characteristics:

typically delay sensitive end-to-end delay delay jitter

loss tolerant: infrequent losses cause minor glitches

antithesis of data, which are loss intolerant but delay tolerant.

Classes of MM applications:

1) stored streaming2) live streaming3) interactive, real-time

Jitter is the variability of packet delays within the same packet stream


Streaming Stored Multimedia

Stored streaming: media stored at source transmitted to client streaming: client playout

begins before all data has arrived timing constraint for still-to-be transmitted

data: in time for playout so collect ~10 sec buffer before playout

starts


Streaming Stored Multimedia: What is it?

1. videorecorded

2. videosent

3. video received,played out at client

Cum

ula

tive

data

streaming: at this time, client playing out early part of video, while server still sending laterpart of video

networkdelay

time


Streaming Stored Multimedia: Interactivity

VCR-like functionality: client can pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect

OK

timing constraint for still-to-be transmitted data: in time for playout


Streaming Live Multimedia

Examples: Internet radio talk show live sporting eventStreaming (as with streaming stored multimedia) playback buffer playback can lag tens of seconds after

transmission timing constraintInteractivity fast forward impossible rewind, pause possible!


Real-Time Interactive Multimedia

end-end delay requirements: audio: < 150 msec good, < 400 msec OK

• includes application-level (packetization) and network delays• higher delays noticeable, impair interactivity

session initialization coordinate IP address, port number, encoding algorithms?

Signaling protocols (SIP, SKYPE)

applications: IP telephony, video conference, distributed interactive worlds


Multimedia Over Today’s InternetTCP/UDP/IP: “best-effort service” no guarantees on delay, loss

Today’s Internet multimedia applications use application-level techniques to mitigate

(as best possible) effects of delay, loss

But you said multimedia apps requiresQoS and level of performance to be

effective!

?? ???

?

? ??

?

?


How should the Internet evolve to better support multimedia?

Integrated services philosophy:

fundamental changes in Internet so that apps can reserve end-to-end bandwidth

requires new, complex software in hosts & routers

Laissez-faire no major changes more bandwidth when

needed content distribution,

application-layer multicast application layer

Differentiated services philosophy:

fewer changes to Internet infrastructure, yet provide 1st and 2nd class service

What’s your opinion?


Chapter 7 outline







Streaming Stored Multimedia

application-level streaming techniques for making the best out of best effort service: client-side buffering use of UDP versus

TCP multiple encodings

of multimedia

jitter removal decompression error concealment graphical user interface

w/ controls for interactivity

Media Player


constant bit rate videotransmission

Cum

ula

tive

data

time

variablenetwork

delay

client videoreception

constant bit rate video playout at client

client playoutdelay

bu

ffere

dvid

eo

Streaming Multimedia: Client Buffering

client-side buffering, playout delay compensate for network-added delay, delay jitter


Streaming Multimedia: UDP or TCP?UDP server sends at rate appropriate for client (oblivious to network congestion !)

often send rate = encoding rate = constant rate then, fill rate = constant rate - packet loss

short playout delay (2-5 seconds) to remove network jitter error recover: time permitting

TCP send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls


User Control of Streaming Media: RTSP

HTTP does not target

multimedia content no commands for fast

forward, etc.RTSP: RFC 2326 client-server

application layer protocol

user control: rewind, fast forward, pause, resume, repositioning, etc…

What it doesn’t do: doesn’t define how

audio/video is encapsulated for streaming over network

doesn’t restrict how streamed media is transported (UDP or TCP possible)

doesn’t specify how media player buffers audio/video


Chapter 7 outline







Internet Phone: Packet Loss and Delay

network loss: IP datagram lost due to network congestion (router buffer overflow)

delay loss: IP datagram arrives too late for playout at receiver delays: processing, queueing in network;

end-system (sender, receiver) delays typical maximum tolerable delay: 400 ms

loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated.


Internet Phone: Fixed Playout Delay

receiver attempts to playout each chunk exactly q msecs after chunk was generated. chunk has time stamp t: play out chunk at

t+q . chunk arrives after t+q: data arrives too

late for playout, data “lost” tradeoff in choosing q:

large q: less packet loss small q: better interactive experience


Fixed Playout Delay

packets

tim e

packetsgenerated

packetsreceived

loss

r

p p '

playout schedulep' - r

playout schedulep - r

• sender generates packets every 20 msec during talk spurt.• first packet received at time r• first playout schedule: begins at p• second playout schedule: begins at p’


Adaptive Playout Delay (1)

packetith receivingafter delay network average of estimated

acketpith for delay network tr

receiverat played is ipacket timethep

receiverby received is ipacket timether

packetith theof timestampt

i

ii

i

i

i

dynamic estimate of average delay at receiver:

)()1( 1 iiii trudud

where u is a fixed constant (e.g., u = .01).

Goal: minimize playout delay, keeping late loss rate low Approach: adaptive playout delay adjustment:

estimate network delay, adjust playout delay at beginning of each talk spurt.

silent periods compressed and elongated. chunks still played out every 20 msec during talk spurt.


Adaptive playout delay (2)

also useful to estimate average deviation of delay, vi :

||)1( 1 iiiii dtruvuv

estimates di , vi calculated for every received packet (but used only at start of talk spurt

for first packet in talk spurt, playout time is:

iiii Kvdtp where K is positive constant

remaining packets in talkspurt are played out periodically


Adaptive Playout (3)

Q: How does receiver determine whether packet is first in a talkspurt?

if no loss, receiver looks at successive timestamps. difference of successive stamps > 20 msec -->talk

spurt begins. with loss possible, receiver must look at both

time stamps and sequence numbers. difference of successive stamps > 20 msec and

sequence numbers without gaps --> talk spurt begins.


Recovery from packet loss (1)

Forward Error Correction (FEC): simple scheme

for every group of n chunks create redundant chunk by exclusive OR-ing n original chunks

send out n+1 chunks, increasing bandwidth by factor 1/n.

can reconstruct original n chunks if at most one lost chunk from n+1 chunks

playout delay: enough time to receive all n+1 packets

tradeoff: increase n, less

bandwidth waste increase n, longer

playout delay increase n, higher

probability that 2 or more chunks will be lost



2nd FEC scheme “piggyback lower quality stream” send lower resolutionaudio stream as redundant information e.g., nominal stream PCM at 64 kbpsand redundant streamGSM at 13 kbps.

whenever there is non-consecutive loss, receiver can conceal the loss. can also append (n-1)st and (n-2)nd low-bit ratechunk



Interleaving chunks divided into smaller units for example, four 5 msec units

per chunk packet contains small units from

different chunks

if packet lost, still have most of every chunk

no redundancy overhead, but increases playout delay


Content distribution networks (CDNs)

Content replication challenging to stream large

files (e.g., video) from single origin server in real time

solution: replicate content at hundreds of servers throughout Internet content downloaded to

CDN servers ahead of time placing content “close” to

user avoids impairments (loss, delay) of sending content over long paths

CDN server typically in edge/access network

origin server in North America

CDN distribution node

CDN serverin S. America CDN server

in Europe

CDN serverin Asia


Content distribution networks (CDNs)

Content replication CDN (e.g., Akamai)

customer is the content provider (e.g., CNN)

CDN replicates customers’ content in CDN servers.

when provider updates content, CDN updates servers

origin server in North America

CDN distribution node

CDN serverin S. America CDN server

in Europe

CDN serverin Asia


CDN example

origin server (www.foo.com) distributes HTML replaces: http://www.foo.com/sports.ruth.gif

with

http://www.cdn.com/www.foo.com/sports/ruth.gif

HTTP request for

www.foo.com/sports/sports.html

DNS query for www.cdn.com

HTTP request for

www.cdn.com/www.foo.com/sports/ruth.gif

1

2

3

origin server

CDN’s authoritative DNS server

CDN server near client

CDN company (cdn.com)

distributes gif files uses its authoritative

DNS server to route redirect requests

client


More about CDNs

routing requests CDN creates a “map”, indicating distances

from leaf ISPs and CDN nodes when query arrives at authoritative DNS

server: server determines ISP from which query originates uses “map” to determine best CDN server

CDN nodes create application-layer overlay network


Chapter 7 outline







Providing Multiple Classes of Service

thus far: making the best of best effort service one-size fits all service model

alternative: multiple classes of service partition traffic into classes network treats different classes of traffic

differently (analogy: VIP service vs regular service)

0111

granularity: differential service among multiple classes, not among individual connections

history: ToS bits


Multiple classes of service: scenario

R1 R2H1

H2

H3

H41.5 Mbps linkR1 output

interface queue


Scenario 1: mixed FTP and audio

Example: 1Mbps IP phone, FTP share 1.5 Mbps link. bursts of FTP can congest router, cause audio loss want to give priority to audio over FTP

packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly

Principle 1

R1 R2


Principles for QOS Guarantees (more) what if applications misbehave (audio sends higher

than declared rate) policing: force source adherence to bandwidth allocations

marking and policing at network edge: similar to ATM UNI (User Network Interface)

provide protection (isolation) for one class from othersPrinciple 2

R1 R2

1.5 Mbps link

1 Mbps phone

packet marking and policing


Principles for QOS Guarantees (more)

Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation

While providing isolation, it is desirable to use resources as efficiently as possible

Principle 3

R1R2

1.5 Mbps link

1 Mbps phone

1 Mbps logical link

0.5 Mbps logical link


Scheduling And Policing Mechanisms

scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of

arrival to queue real-world example? discard policy: if packet arrives to full queue: who to

discard?• Tail drop: drop arriving packet• priority: drop/remove on priority basis• random: drop/remove randomly


Scheduling Policies: more

Priority scheduling: transmit highest priority queued packet

multiple classes, with different priorities class may depend on marking or other header info, e.g.

IP source/dest, port numbers, etc.. Real world example?


Scheduling Policies: still moreround robin scheduling: multiple classes cyclically scan class queues, serving one from each class (if available) real world example?


Scheduling Policies: still more

Weighted Fair Queuing: generalized Round Robin each class gets weighted amount of service in

each cycle real-world example?


Policing Mechanisms

Goal: limit traffic to not exceed declared parameters

Three common-used criteria of traffic rate: (Long term) Average Rate: how many packets can

be sent per unit time (in the long run) crucial question: what is the interval length: 100 packets

per sec or 6000 packets per min have same average!

Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 15000 ppm peak rate

(Max.) Burst Size: max. number of packets sent consecutively (with no intervening idle)


Policing Mechanisms

Token Bucket: limit input to specified Burst Size and Average Rate.

bucket can hold b tokens tokens generated at rate r token/sec unless

bucket full over interval of length t: number of packets

admitted less than or equal to (r t + b).


Edge router: per-flow traffic

management

marks packets as in-profile and out-profile

Core router: per class traffic management buffering and scheduling

based on marking at edge preference given to in-profile

packets

Diffserv Architecture

scheduling

...

r

b

marking


Edge-router Packet Marking

class-based marking: packets of different classes marked differently

intra-class marking: conforming portion of flow marked differently than non-conforming one

profile: pre-negotiated rate A, bucket size B packet marking at edge based on per-flow profile

Possible usage of marking:

User packets

Rate A

B


Classification and Conditioning

Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6

6 bits used for Differentiated Service Code Point (DSCP) and determine per hop behavior (PHB) that the packet will receive

2 bits are currently unused


Classification and Conditioning

may be desirable to limit traffic injection rate of some class:

user declares traffic profile (e.g., rate, burst size)

traffic metered, shaped if non-conforming


Forwarding (PHB)

PHB result in a different observable (measurable) forwarding performance behavior

PHB does not specify what mechanisms to use to ensure required PHB performance behavior

Examples: Class A gets x% of outgoing link bandwidth over time

intervals of a specified length Class A packets leave first before packets from class

B


Forwarding (PHB)

PHBs being developed: Expedited Forwarding: pkt departure rate of a

class equals or exceeds specified rate logical link with a minimum guaranteed rate

Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions


Chapter 7 outline





7.5 providing QoS guarantees: IntServ, RSVP


Principles for QOS Guarantees (more)

Basic fact of life: can not support traffic demands beyond link capacity

Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs

Principle 4

R1R2

1.5 Mbps link

1 Mbps phone

1 Mbps phone


QoS guarantee scenario

Resource reservation call setup, signaling (RSVP) traffic, QoS declaration per-element admission control

QoS-sensitive scheduling (e.g.,

WFQ)

request/reply


IETF Integrated Services

architecture for providing QOS guarantees in IP networks for individual application sessions

resource reservation: routers maintain state info (like VC) of allocated resources, per flow QoS req’s

admit/deny new call setup requests:

Question: can newly arriving flow be admitted with performance guarantees while not violating QoS guarantees made to already admitted flows?


Call Admission

Arriving session must : declare its QOS requirement

R-spec: defines the QOS being requested characterize traffic it will send into network

T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and T-

spec to routers (where reservation is required) RSVP


Intserv QoS: Service models [rfc2211, rfc 2212]

Guaranteed service: worst case traffic arrival:

leaky-bucket-policed source simple (mathematically

provable) bound on delay [Parekh 1992, Cruz 1988]

Controlled load service: "a quality of service

closely approximating the QoS that same flow would receive from an unloaded network element."

WFQ

token rate, r

bucket size, b

per-flowrate, R

D = b/Rmax

arrivingtraffic


Signaling in the Internet

connectionless (stateless)

traffic forwarding by IP routers

best effort service

no network signaling protocols

needed in initial IP design

+ =

New requirement: reserve resources along end-to-end path (end system, routers) for QoS for multimedia applications

RSVP: Resource Reservation Protocol [RFC 2205] “ … allow users to communicate requirements to

network in robust and efficient way.” i.e., signaling !


RSVP Design Goals

1. accommodate heterogeneous receivers (different bandwidth along paths)

2. accommodate different applications with different resource requirements

3. make multicast a first class service, with adaptation to multicast group membership

4. leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes

5. control protocol overhead to grow (at worst) linear in # receivers

6. modular design for heterogeneous underlying technologies


RSVP: does not…

specify how resources are to be reserved rather: a mechanism for communicating needs

determine routes packets will take that’s the job of routing protocols signaling decoupled from routing


RSVP: overview of operation

senders, receiver join a multicast group done outside of RSVP senders need not join group

sender-to-network signaling path message: make sender presence known to routers path teardown: delete sender’s path state from routers

receiver-to-network signaling reservation message: reserve resources from sender(s)

to receiver reservation teardown: remove receiver reservations

network-to-end-system signaling path error reservation error


Chapter 7: Summary

Principles classify multimedia applications making the best of best effort service

Protocols and Architectures specific protocols for best-effort mechanisms for providing QoS architectures for QoS

multiple classes of service QoS guarantees, admission control

11: Multimedia Networking 11-1 Chapter 11 Multimedia Networking A note on the use of these ppt slides: We’re making these slides freely available to all.

Documents

multimedia networking11

multimedia applications

multimedia networkinga

stored audio

ppt slides

best effort service7

transmitted data

unaltered form